- 06 12月, 2010 3 次提交
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由 Clemens Ladisch 提交于
The "Front Panel" switch on the Xonar D1/DX actually switches only the output direction, so mark it appropriately. The front panel microphone is controlled by the FMIC2MIC bit of the CM9780. It was unconditionally enabled on the D1/DX and never set on the ST(X); add a control for it. Selecting the front panel microphone as source does not actually disable the microphone jack, but this is bug-compatible with the Windows driver, and users rely on it. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Clemens Ladisch 提交于
The GPIO bit that enables analog output on the Xonar HDAV1.3 also disables the HDMI audio output, so we better add a switch for it. Hopefully, this is sufficient to make the HDMI output work. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Clemens Ladisch 提交于
Initialize the configuration of some unknown GPIO output bits (that might not be used at all) to be the same as in the Windows driver, just to be sure. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 01 12月, 2010 1 次提交
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由 Florian Faber 提交于
Add support for the RME HDSP RPM IO box. Changes have been made in the identification of the IO box and the neccessary controls have been added. Signed-off-by: NFlorian Faber <faberman@linuxproaudio.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 24 11月, 2010 1 次提交
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由 Kay Sievers 提交于
If CONFIG_SND_DYNAMIC_MINORS is used, assign /dev/snd/seq and /dev/snd/timer the usual static minors, and export specific module aliases to generate udev module on-demand loading instructions: $ cat /lib/modules/2.6.33.4-smp/modules.devname # Device nodes to trigger on-demand module loading. microcode cpu/microcode c10:184 fuse fuse c10:229 ppp_generic ppp c108:0 tun net/tun c10:200 uinput uinput c10:223 dm_mod mapper/control c10:236 snd_timer snd/timer c116:33 snd_seq snd/seq c116:1 The last two lines instruct udev to create device nodes, even when the modules are not loaded at that time. As soon as userspace accesses any of these nodes, the in-kernel module-loader will load the module, and the device can be used. The header file minor calculation needed to be simplified to make __stringify() (supports only two indirections) in the MODULE_ALIAS macro work. This is part of systemd's effort to get rid of unconditional module load instructions and needless init scripts. Cc: Lennart Poettering <lennart@poettering.net> Signed-off-by: NKay Sievers <kay.sievers@vrfy.org> Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 22 11月, 2010 16 次提交
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由 Clemens Ladisch 提交于
Increase the default timer limit so that snd-hrtimer.ko can be automatically loaded when needed, e.g., when used as the default sequencer timer. This replaces the check for the obsolete CONFIG_SND_HPET. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Clemens Ladisch 提交于
Add a lightweight condition on top of the xrun checking so that we can avoid the division when the application is calling the update function often enough. Suggested-by: NJaroslav Kysela <perex@perex.cz> Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Clemens Ladisch 提交于
When period wakeups are disabled, successive calls to the pointer update function do not have a maximum allowed distance, so xruns cannot be detected with the pointer value only. To detect xruns, compare the actually elapsed time with the time that should have theoretically elapsed since the last update. When the hardware pointer has wrapped around due to an xrun, the actually elapsed time will be too big by about hw_ptr_buffer_jiffies. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Clemens Ladisch 提交于
Allow disabling period wakeup interrupts for all PCM streams. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Clemens Ladisch 提交于
Allow disabling period wakeup interrupts for HDA PCM streams. Signed-off-by: NPierre-Louis Bossart <pierre-louis.bossart@intel.com> Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Clemens Ladisch 提交于
This patch allows to disable period interrupts which are not needed when the application relies on a system timer to wake-up and refill the ring buffer. The behavior of the driver is left unchanged, and interrupts are only disabled if the application requests this configuration. The behavior in case of underruns is slightly different, instead of being detected during the period interrupts the underruns are detected when the application calls snd_pcm_update_avail, which in turns forces a refresh of the hw pointer and shows the buffer is empty. More specifically this patch makes a lot of sense when PulseAudio relies on timer-based scheduling to access audio devices such as HDAudio or Intel SST. Disabling interrupts removes two unwanted wake-ups due to period elapsed events in low-power playback modes. It also simplifies PulseAudio voice modules used for speech calls. To quote Lennart "This patch looks very interesting and desirable. This is something have long been waiting for." Support for this in hardware drivers is optional. Signed-off-by: NPierre-Louis Bossart <pierre-louis.bossart@intel.com> Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
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由 Daniel T Chen 提交于
BugLink: https://launchpad.net/bugs/677652 The original reporter states that, in 2.6.35, headphones do not appear to work, nor does inserting them mute the A52J's onboard speakers. Upon inspecting the codec dump, it appears that the newly committed hp-laptop quirk will suffice to enable this basic functionality. Testing was done with an alsa-driver build from 2010-11-21. Reported-and-tested-by: Joan Creus Cc: <stable@kernel.org> [2.6.35+] Signed-off-by: NDaniel T Chen <crimsun@ubuntu.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Vasiliy Kulikov 提交于
After clk_get() pclk is checked second time instead of sample_clk check. Signed-off-by: NVasiliy Kulikov <segoon@openwall.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Andreas Mohr 提交于
. Fix PulseAudio "ALSA driver bug" issue (if we have two alternated areas within a 64k DMA buffer, then max period size should obviously be 32k only). Back references: http://pulseaudio.org/wiki/AlsaIssues http://fedoraproject.org/wiki/Features/GlitchFreeAudio . In stop timer function, need to supply ACK in the timer control byte. . Minor log output correction When I did my first PA testing recently, the period size bug resulted in quite precisely observeable half-period-based playback distortion. PA-based operation is quite a bit more underrun-prone (despite its zero-copy optimizations etc.) than raw ALSA with this rather spartan sound hardware implementation on my puny Athlon. Note that even with this patch, azt3328 still doesn't work for both cases yet, PA tsched=0 and tsched (on tsched=0 it will playback tiny fragments of periods, leading to tiny stuttering sounds with some pauses in between, whereas with timer-scheduled operation playback works fine - minus some quite increased underrun trouble on PA vs. ALSA, that is). Signed-off-by: NAndreas Mohr <andi@lisas.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Daniel T Chen 提交于
BugLink: https://launchpad.net/bugs/677830 The original reporter states that the subwoofer does not mute when inserting headphones. We need an entry for his machine's SSID in the subwoofer pin fixup list, so add it there (verified using hda_analyzer). Reported-and-tested-by: i-NoD Cc: <stable@kernel.org> Signed-off-by: NDaniel T Chen <crimsun@ubuntu.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 David Henningsson 提交于
BugLink: http://launchpad.net/bugs/669092 ALC887 does not have any volume control ability on the mixer NIDs, so put the volume controls on the dac NIDs instead. Without this patch, ALC887 users cannot use alsamixer at all. Cc: stable@kernel.org Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Joe Perches 提交于
Signed-off-by: NJoe Perches <joe@perches.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Joe Perches 提交于
Signed-off-by: NJoe Perches <joe@perches.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Joe Perches 提交于
Using %pR standardizes the struct resource output. Signed-off-by: NJoe Perches <joe@perches.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Daniel T Chen 提交于
BugLink: https://launchpad.net/bugs/669279 The original reporter states: "The Master mixer does not change the volume from the headphone output (which is affected by the headphone mixer). Instead it only seems to control the on-board speaker volume. This confuses PulseAudio greatly as the Master channel is merged into the volume mix." Fix this symptom by applying the hp_only quirk for the reporter's SSID. The fix is applicable to all stable kernels. Reported-and-tested-by: NBen Gamari <bgamari@gmail.com> Cc: <stable@kernel.org> [2.6.32+] Signed-off-by: NDaniel T Chen <crimsun@ubuntu.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 11 11月, 2010 6 次提交
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由 Peter Rosin 提交于
The Atmel SSC can divide by even numbers, not only powers of two. Signed-off-by: NPeter Rosin <peda@axentia.se> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Julia Lawall 提交于
In each function, the value apcm is stored in the private_data field of runtime. At the same time the function ct_atc_pcm_free_substream is stored in the private_free field of the same structure. ct_atc_pcm_free_substream dereferences and ultimately frees the value in the private_data field. But each function can exit in an error case with apcm having been freed, in which case a subsequent call to the private_free function would perform a dereference after free. On the other hand, if the private_free field is not initialized, it is NULL, and not invoked (see snd_pcm_detach_substream in sound/core/pcm.c). To avoid the introduction of a dangling pointer, the initializations of the private_data and private_free fields are moved to the end of the function, past any possible free of apcm. This is safe because the previous calls to snd_pcm_hw_constraint_integer and snd_pcm_hw_constraint_minmax, which take runtime as an argument, do not refer to either of these fields. In each function, there is one error case where apcm needs to be freed, and a call to kfree is added. The sematic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression e,e1,e2,e3; identifier f,free1,free2; expression a; @@ *e->f = a ... when != e->f = e1 when any if (...) { ... when != free1(...,e,...) when != e->f = e2 * kfree(a) ... when != free2(...,e,...) when != e->f = e3 } // </smpl> Signed-off-by: NJulia Lawall <julia@diku.dk> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Florian Fainelli 提交于
If the platform already provides a definition for these accessors do not redefine them. The warning was caught on MIPS. Signed-off-by: NFlorian Fainelli <florian@openwrt.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 David Henningsson 提交于
BugLink: http://launchpad.net/bugs/673075 According to the datasheet of 92HD87B, there is a digital mic at nid 0x11, so enable it in order to be able to use the mic. Cc: stable@kernel.org Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Jesper Juhl 提交于
The [vk][cmz]alloc(_node) family of functions return void pointers which it's completely unnecessary/pointless to cast to other pointer types since that happens implicitly. This patch removes such casts from sound/oss/ Signed-off-by: NJesper Juhl <jj@chaosbits.net> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Joe Perches 提交于
Signed-off-by: NJoe Perches <joe@perches.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 03 11月, 2010 9 次提交
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由 Takashi Iwai 提交于
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由 Jarkko Nikula 提交于
Patch "ASoC: tpa6130a2: Fix unbalanced regulator disables" introduced a compiler warning "‘ret’ may be used uninitialized in this function". Initialize ret to zero to get rid of it and making sure that the function does not return any random error code when the code is falling through. Signed-off-by: NJarkko Nikula <jhnikula@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Clemens Ladisch 提交于
Add support for the TempoTec/MediaTek HiFier Serenade sound card. The PCI ID was already there, but the driver handled it like the Fantasia model, which resulted in a dummy recording device. As a stereo output-only card, this model is to be handled exactly like the HG2PCI. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Clemens Ladisch 提交于
Sort the PCI IDs so that they make logical sense. Also move the card name comments into this list because the model symbols should be (more) self-explanationary. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Merge branch 'for-2.6.37' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into fix/asoc
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由 Clemens Ladisch 提交于
Add support for the Kuroutoshikou CMI8787-HG2PCI sound card. [replaced non-latin letters in the patch by tiwai] Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Clemens Ladisch 提交于
The snd-hifier driver contains more duplicated code than model-specific code, so it does not make sense for it to be a separate driver. Handling the two-channel output restriction can be easily done in the generic driver. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
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由 Edgar (gimli) Hucek 提交于
This patch add support for the MacBookAir3,1 and MacBookAir3,2 to the alsa sound system. Signed-off-by: NEdgar (gimli) Hucek <gimli@dark-green.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 02 11月, 2010 4 次提交
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由 Mark Brown 提交于
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由 Eric Miao 提交于
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mandar Joshi 提交于
This patch adds support for Power/Status LED on Creative USB X-Fi S51. There is just one LED on the device. The LED can either be On or it can be set to Blink. There doesn't seem to be a way to switch it off. The control message to change LED status is similar to that of audigy2nx except that the index is to be set to 0 and value is 1 for Blink and 0 for On. The 'Power LED' control in alsamixer when muted will cause the LED to Blink continuously. When unmuted the LED will stay On. The Creative driver under Windows sets the LED to blink whenever audio is muted. This LED can be treated as the CMSS LED but I figured since there is just one LED, it should be treated as the Power LED. Is that alright? I've also changed the comment "Usb X-Fi" to "Usb X-Fi S51" as there are other external X-Fi devices from Creative like Usb X-Fi Go and Xmod. The volume knob and LED support patch doesn't apply to them. Signed-off-by: NMandar Joshi <emailmandar@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Jesper Juhl 提交于
I noticed that sound/pci/asihpi/hpicmn.c::hpi_alloc_control_cache() does not check the return value from kmalloc(), which may fail. If kmalloc() fails we'll dereference a null pointer and things will go bad fast. There are two memory allocations in that function and there's also the problem that the first may succeed and the second may fail and nothing is done about that either which will also go wrong down the line. Signed-off-by: NJesper Juhl <jj@chaosbits.net> Acked-by: NEliot Blennerhassett <linux@audioscience.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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